In telecommunications systems transmitting digital speech, a speech signal is usually subjected to two coding operations: speech coding and channel coding.
Speech coding comprises speech encoding performed in the transmitter by a speech encoder and speech decoding performed in the receiver by a speech decoder. The speech encoder in the transmitter compresses a speech signal so that the number of bits used for representing the speech signal per a unit of time is reduced, whereby less transmission capacity is required for transmitting the speech signal. The speech decoder in the receiver performs a reverse operation and synthesizes the speech signal from the bits generated by the speech encoder. However, the speech synthesized in the receiver is not identical with the original speech compressed by the speech encoder; the original speech has changed more or less as a result of the speech coding. In general, the more the speech is compressed in the speech coding, the more its quality deteriorates during the coding. In the pan-European GSM mobile communication system (Global System for Mobile Communication), for example, the speech encoder of a full-rate traffic channel compresses a speech signal to a transmission rate of 13 kbit/s. The speech synthesized by the corresponding speech decoder is clearly of a poorer quality than the speech transmitted by, for instance, a public switched telephone network (PSTN).
Thus, when a speech coding method is selected, a compromise must be made between the quality offered by the method and the transmission capacity required by it. Another factor to be considered in the selection is the complexity of the implementation of the speech coding method: the quality of speech can usually be improved without increasing the transmission rate if higher requirements for the method as regards calculation capacity and thereby also higher costs of the implementation are allowed. On account of the continuous development of speech coding methods and the implementation techniques, more and more advanced methods are available for speech transmission in the existing telecommunications systems. After the development of the method employed in the GSM, speech coding technology has advanced to such an extent that, as compared with the above-mentioned 13 kbit/s speech coding method, a higher quality of speech can now be achieved at a much lower transmission rate, e.g. 8 kbit/s.
Channel coding comprises channel encoding performed in the transmitter by a channel encoder, and channel decoding performed in the receiver by a channel decoder. The purpose of channel coding is to protect speech coding bits to be transmitted against errors occurring in the transmission channel. Channel coding can either be used for merely detecting whether the transmission has caused any errors in the speech coding bits without any possibility of correcting them, or it may be capable of correcting errors caused by the transmission, provided that the number of errors does not exceed a given maximum, which is dependent on the channel coding method.
The selection of the channel coding method employed depends on the quality of the transmission channel. In fixed transmission networks the probability of errors is often very low, wherefore not much channel coding is required, whereas in wireless networks such as mobile telephone networks the probability of errors in the transmission channels is often very high, and the channel coding method employed has a significant effect on the resulting quality of speech. Mobile telephone networks usually employ both error-detecting and error-correcting channel coding methods concurrently.
Channel coding is based on the use of error check bits, also called channel coding bits, added to the speech encoding bits. Bits produced by the speech encoder of the transmitter are supplied to a channel encoder, which adds a number of error check bits to them. In the above-mentioned GSM full-rate transmission channel, for example, error check bits with a transmission rate of 9.8 kbit/s are added to speech coding bits of 13 kbit/s on the transmission channel, whereby the total transmission rate of the speech signal on the channel will be 22.8 kbit/s. The channel decoder decodes the channel encoding in the receiver in such a way that only the 13 kbit/s bit stream produced by the speech encoder is applied to the speech decoder. During channel decoding, the channel decoder detects and/or corrects errors that have occurred on the channel as far as such error correction is possible.
Speech coding and channel coding are closely connected with each other in telecommunications systems transmitting speech. The significance of the bits produced by the speech encoder for the quality of speech generally varies so that in some cases one error in an important bit may cause audible noise in the synthesized voice, whereas a larger number of errors in less important bits may be almost imperceptible. How big the differences between the importance of speech coding bits are depends essentially on the speech coding method employed; however, at least small differences can be found in most methods. When a speech transmission method is developed for a telecommunications system, channel coding is therefore usually designed together with speech coding in such a manner that the bits that are the most important for the quality of speech are protected more carefully than less important bits. On a full-rate channel of the GSM system, for instance, the bits produced by the speech encoder are divided into three different categories according to their importance. The most important category is protected in channel coding with both an error-detecting and an error-correcting code; the second most important category is protected only with an error-detecting code; the least important category is not protected at all in channel coding.
Although the speech coding and channel coding are closely connected, there are often considerable differences in their implementation in digital mobile telephone networks. The GSM system may once again be used as an example. Speech encoding and speech decoding are typically carried out by means of software, using a digital signal processor. This applies both to terminal equipment (telephones) and to network elements. Channel coding may also be performed by means of software, but often a separate integrated circuit is designed for this purpose, especially at the network end. Thus, changing of the speech coding method requires often merely a new signal processing program, whereas changing of the channel coding method may require equipment changes.
In addition to the way they are implemented, these two codings, speech coding and channel coding, may differ in their physical locations at the network end of a mobile telephone system. In the GSM system, for example, channel coding in the network is performed in a base station, while speech coding is performed in a separate transcoder unit, which may be remote from the base station, and even if it is located at the base station, it is a completely separate unit. Because of the separate locations, any changes in the transmission rates of the channel coding and speech coding will also entail changes in the connections between the different network elements.
In view of the different ways in which speech and channel coding are performed and their separate locations, it would be clearly more advantageous if the quality of speech could be improved in an existing system merely by changing the speech coding. As the channel coding is, however, usually designed particularly for the speech coding of the existing system, and as the new speech coding method should use exactly the same transmission rate as the original speech coding method of the system, methods for adapting new speech coding methods for existing telecommunications systems have not been disclosed previously.
FIG. 1A and 1B are block diagrams illustrating a transmitter and a receiver of a prior art telecommunications system. In the transmitter shown in FIG. 1A, a speech signal 100 is supplied to a speech encoder 101, which on the basis of the signal generates compressed speech coding bits having a transmission rate of S kbit/s. These speech coding bits are supplied to a channel encoder 102, where error check bits are added to them, which results in a total transmission rate of S+C kbit/s. This bit stream 103 is transmitted over the transmission channel to the receiver shown in FIG. 1B. In the receiver of FIG. 1B, the bit stream 104 received from the transmitter is at first supplied to a channel decoder 105, which decodes the channel encoding and transmits the speech coding bits thus obtained to a speech decoder 106; the transmission rate of the speech coding bits is again S kbit/s. The speech decoder synthesizes a digital speech signal 107. The telecommunications systems of the prior art thus employ only one speech encoding method and a corresponding channel coding method. Such telecommunications systems include, for example, all the commonest digital mobile telephone systems.
The prior art systems also include systems in which two different speech coding methods are used in such a manner that a separate channel coding method corresponds to each speech coding method, and in which the total transmission rate obtained as a result of the speech and channel coding is different in these two methods. An example of such a system is the GSM mobile telephone system, in which full-rate and half-rate traffic channels are specified.
There are also known solutions in which transmitters and receivers according to FIG. 1A and 1B are connected in parallel so that the system that is formed comprises several different speech encoding methods, each of which has a corresponding channel coding method. The speech coding methods used in such a system can operate at different transmission rates, wherefore the channel coding methods corresponding to them are also mutually independent and operate at different transmission rates.